Method of guaranteeing jitter upper bound for a network without time-synchronization

ABSTRACT

In a method of guaranteeing a jitter upper bound for a network without time-synchronization, which guarantees a jitter upper bound for a flow that is transmitted from a source to a destination through a network, the network guarantees a latency upper bound of the flow, a buffer located between the network and the destination holds a packet of the flow for a predetermined buffer holding interval and then outputs, and the jitter upper bound is set to an arbitrary value including 0 (zero).

CROSS-REFERENCE TO RELATED APPLICATION

This application claims priority of Korean Patent Application No.10-2021-0016065, filed on Feb. 4, 2021, in the KIPO (Korean IntellectualProperty Office), the disclosure of which is incorporated hereinentirely by reference.

BACKGROUND OF THE INVENTION Field of the Invention

The present invention relates to a method of guaranteeing a jitter upperbound for a network without time-synchronization, and more particularly,to a method of guaranteeing any jitter upper bound including zero jitterwithout time-synchronization between nodes included in a network.

Description of the Related Art

The demands for deterministic network services are getting significant,and in particular, the demands are strong in closed small networks suchas in-car networks or smart factory networks, in which the latency andjitter requirements are clearly defined.

Typical control cycles in automated factories are 100 to 1000 ms with ajitter less than 1 ms, while in robotics, the control cycles may be lessthan 1 ms with a jitter less than 1 μs. The On-Board Software ReferenceArchitecture Network Communication Specification (OSRA-NET), theEuropean standard for satellite network, defines seven communicationclasses and their jitter upper bounds as well as the latency upperbounds as shown in Table 1 below. A class 6 flow in OSRA-NET, forexample, requires maximum latency of 10 ms and maximum jitter of 2 ms,while having more than 100 Mbps data rate.

TABLE 1 Frequency Data rate Max Jitter Max Latency Class (Hz) (bps) (ms)(ms) 1 0.1-1   100-10K 10 10 2-a 8-10 <1M 5-10 10 2-b 8-10 <1M 5-10 10 38-10  <250K 10 10 4 0.1-1   >100M  <100 <100 5-a 10-1K  <3M 0.5-1   0.55-b 10-1K  <3M 0.5-1   0.5 6 1-10 >100M  2 10 7 1-10  100-1K 1 2

Common public radio interface (CPRI) in 5G network is another example ofsmall networks with stringent latency and jitter requirements. CPRI is apacket-based constant-bit-rate protocol developed to transport databetween the remote radio heads (RRHs) and baseband units (BBUs) pool,and the CPRI network is usually a simple single-hop network. Whileachieving the throughput of 10 Gbps as required in the fronthaulnetworks, CPRI is expected to maintain the maximum end-to-end latenciesless than 100 to 250 μs, jitter within 65 to 100 ns and BER (bit errorrate) less than 10⁻¹², otherwise performance of the networks degradessignificantly.

Meanwhile, there are also emerging applications that require latency andjitter upper bound in larger scale networks. Machine-to-machinecommunication for cloudified industrial and robotic automation involvesmoderate-to-large scale networks. This type of communications requiresvery fine-grained timing accuracy for the dispersion of control commandsand for the collection of telemetry data over a wide area. ITU-T SG-13has defined such services to support critical grade reliability andextremely low as well as highly precise latency. This is because someindustrial controllers require very precise synchronization and spacingof telemetry streams and control data.

The problem of guaranteeing the jitter upper bound in arbitrary sizednetworks with any type of topology, with random dynamic input trafficmay be considered. The jitter is defined to be the latency difference oftwo packets within a flow, not a difference from a clock signal or froman average latency, as it is clearly summarized in RFC 3393.

In large scale networks, the end-nodes join/leave frequently, and theflows are dynamically generated and terminated. Achieving satisfactorydeterministic performance in such an environment would be challenging.

TSN task group has standardized multiple components for jitter-sensitiveservices. Among them, the 802.1 Qbv time sensitive queues (also known asTime Aware Shaper, TAS) and 802.1 Qch cyclic queuing and forwarding(CQF) are built for jitter minimization as well as latency guarantee.

TAS defines the “gate” to each queue. The gates are either open orclosed on a time slot. Multiple gates may open simultaneously. In such acase, the strict priority scheduling applies to the open queues. Acentral network controller (CNC) determines which gates and when toopen. CQF function coordinates the slot openings in cascading nodes,such that flows traversing the nodes experience the minimum latency.Based on these functions, TSN supports a model for deterministic packetservices. These functions collectively are celled the TSN synchronousapproach. The TSN synchronous approach requires three major efforts; 1)the time synchronization along the nodes in the network, 2) the slotscheduling and coordination by a central control device, and 3)feasibility test for flow admission control. These requirementssignificantly impact the scalability of a network.

The slot scheduling is a time and resource consuming task, and as thenumber of flows increases, a greater burden is placed on these limitedresources. Also, for dynamic environments, runtime reconfigurations arenecessary and scheduling becomes harder. In addition, timesynchronization also causes too much overhead, and there is a limit insynchronization accuracy.

With all the efforts until now, however, there is no system that canguarantee a jitter upper bound, including zero jitter, in generalnetworks with arbitrary traffic. For a limited topology, such as asingle-hop CPRI network, there has been research into achieving zerojitter with packet-level slot scheduling methods in lightly utilizedenvironments. This work shows that even for the smallest networks withlightly loaded traffic, it is necessary to micro-manage the slots withthe size of a single packet transmission, and to reorder slotsdynamically, in order for achieving zero jitter.

Therefore, in the art, there is a need for a method of guaranteeing ajitter upper bound even in a large-scale or dynamic network.

SUMMARY OF THE INVENTION

In order to solve the above problems, an embodiment of the presentdisclosure provides a method of guaranteeing a jitter upper bound for adeterministic network without time-synchronization.

In the method of guaranteeing a jitter upper bound for a deterministicnetwork without time-synchronization, which guarantees a jitter upperbound for a flow that is transmitted from a source to a destinationthrough a network, the network guarantees a latency upper bound of theflow, a buffer located between the network and the destination holds apacket of the flow for a predetermined buffer holding interval and thenoutputs, and the jitter upper bound is set to an arbitrary valueincluding 0 (zero).

In addition, the solution to the above problem does not list allfeatures of the present disclosure. Various features of the presentdisclosure and advantages and effects thereof may be understood in moredetail with reference to the following specific embodiments.

According to an embodiment of the present disclosure, it is possible toguarantee a desired jitter upper bound, even zero jitter.

That is, it is possible to guarantee a desired jitter upper boundwithout tremendous effort of the prior art for time-synchronization,slot allocation and coordination between nodes. Accordingly, the jitterupper bound may be guaranteed for dynamic networks of medium or largersizes.

BRIEF DESCRIPTION OF THE DRAWINGS

The above and other features and advantages will become more apparent tothose of ordinary skill in the art by describing in detail exemplaryembodiments with reference to the attached drawings, in which:

FIG. 1 is a diagram showing the overall architecture of a framework fora method of guaranteeing a jitter upper bound for a network withouttime-synchronization according to an embodiment of the presentdisclosure.

FIG. 2 is a diagram showing the relationship between arrival, departureand buffer-out instances of packet of a flow under observation.

FIG. 3 is a diagram showing an example of a network for performanceverification of the present disclosure.

FIG. 4 is a diagram showing a latency distribution obtained from flowsimulation of the C&C using end-to-end latency upper bound and threedifferent frameworks.

FIG. 5 is a diagram showing end-to-end buffering latency of all packetsin two arbitrary sample C&C flow observed during the simulation.

In the following description, the same or similar elements are labeledwith the same or similar reference numbers.

DETAILED DESCRIPTION

The present invention now will be described more fully hereinafter withreference to the accompanying drawings, in which embodiments of theinvention are shown. This invention may, however, be embodied in manydifferent forms and should not be construed as limited to theembodiments set forth herein. Rather, these embodiments are provided sothat this disclosure will be thorough and complete, and will fullyconvey the scope of the invention to those skilled in the art.

The terminology used herein is for the purpose of describing particularembodiments only and is not intended to be limiting of the invention. Asused herein, the singular forms “a”, “an” and “the” are intended toinclude the plural forms as well, unless the context clearly indicatesotherwise. It will be further understood that the terms “includes”,“comprises” and/or “comprising,” when used in this specification,specify the presence of stated features, integers, steps, operations,elements, and/or components, but do not preclude the presence oraddition of one or more other features, integers, steps, operations,elements, components, and/or groups thereof. In addition, a term such asa “unit”, a “module”, a “block” or like, when used in the specification,represents a unit that processes at least one function or operation, andthe unit or the like may be implemented by hardware or software or acombination of hardware and software.

Reference herein to a layer formed “on” a substrate or other layerrefers to a layer formed directly on top of the substrate or other layeror to an intermediate layer or intermediate layers formed on thesubstrate or other layer. It will also be understood by those skilled inthe art that structures or shapes that are “adjacent” to otherstructures or shapes may have portions that overlap or are disposedbelow the adjacent features.

In this specification, the relative terms, such as “below”, “above”,“upper”, “lower”, “horizontal”, and “vertical”, may be used to describethe relationship of one component, layer, or region to anothercomponent, layer, or region, as shown in the accompanying drawings. Itis to be understood that these terms are intended to encompass not onlythe directions indicated in the figures, but also the other directionsof the elements.

Unless otherwise defined, all terms (including technical and scientificterms) used herein have the same meaning as commonly understood by oneof ordinary skill in the art to which this invention belongs. It will befurther understood that terms, such as those defined in commonly useddictionaries, should be interpreted as having a meaning that isconsistent with their meaning in the context of the relevant art andwill not be interpreted in an idealized or overly formal sense unlessexpressly so defined herein.

Preferred embodiments will now be described more fully hereinafter withreference to the accompanying drawings. However, they may be embodied indifferent forms and should not be construed as limited to theembodiments set forth herein. Rather, these embodiments are provided sothat this disclosure will be thorough and complete, and will fullyconvey the scope of the disclosure to those skilled in the art.

The core idea of the present disclosure is to buffer a first packet of aflow for an appropriate time interval based on a latency upper boundprovided by a network, and to buffer all subsequent packets of the flowso that inter-arrival intervals thereof become similar tointer-departure intervals.

In the prior art, time-stamp is generally used for latency/jittermeasurement or clock synchronization, but in the present disclosure, itis used as a tool for jitter upper bound guarantee.

Hereinafter, a method of guaranteeing a jitter upper bound for adeterministic network without time-synchronization according to anembodiment of the present disclosure will be described.

FIG. 1 is a diagram showing the overall architecture of a framework fora method of guaranteeing a jitter upper bound for a network withouttime-synchronization according to an embodiment of the presentdisclosure.

In the present disclosure, time-synchronization or slot allocation taskbetween nodes included in the network is not required, and the followingthree components are required:

1) a network 10 that guarantees latency upper bounds;

2) a buffer 20 before a destination, which can hold the packets destinedfor the destination for a predetermined interval; and

3) packets with relative time-stamps inscribed with the clock of thesource, or of the network ingress interface.

The relative time-stamp means that it is not needed to synchronize thesource of the traffic recording a time-stamp with another node. In thepresent disclosure, there is a mechanism for time-stamping a departureinstance from the source or an arrival instance (a_(n)) to the networkto the packet. For example, the time-stamp function of TCP or RTP may beused. Alternatively, a time-stamp implemented in a lower layer, forexample a MAC layer, may also be used. In addition, there is no need toshare a synchronized clock between the source and the destination, andit is sufficient to know the difference between time-stamps, namelyinformation about the relative arrival instance.

Also, the destination may be a small-sized network with a synchronizednetwork, like a TSN synchronous network.

The network 10 is irrelevant to any size, topology or input pattern aslong as it guarantees the latency upper bound (U) of the flow. Inaddition, the network 10 provides a lower latency bound (W) to the flow,and the lower latency bound (W) may be caused by, for example, atransmission and propagation delay within the network.

The time when an n^(th) packet of the flow is input to the network 10 iscalled an arrival instance (a_(n)). The time when the n^(th) packet isoutput from the network 10 is called a departure instance (b_(n)). Forexample, a₁ and b₁ are arrival and departure instances of the firstpacket of the flow, respectively. It is assumed that the bufferingparameter m is a value between W and U, that is, W≤m≤U.

The buffer 20 may hold packets of the flow according to predefinedintervals. To determine the buffer holding interval, a time-stamp ineach packet may be used as described later.

It may be assumed that the buffer 20 is able to support as many as thenumber of flows destined for the destination. In addition, if the buffer20 is not suitable to be placed within an end station, a bufferingfunction may be added at the boundary of the network 10.

FIG. 1 shows only a single flow between a source and a destination,where the arrival instance (a_(n)), the departure instance (b_(n)) andthe buffer-out instance (c_(n)) of the n^(th) packet of the flow aredepicted. Here, end-to-end latency and end-to-end buffering latency maybe defined as (b_(n)−a_(n)) and (c_(n)−a_(n)) , respectively.

The basic rule for determining the buffer holding interval is asfollows.

-   -   The buffer 20 holds the first packet for an interval (m−W), for        m, W≤m≤U. The buffer-out instance of the first packet c₁ is        (b_(1+m−W).)    -   The buffer 20 holds the n^(th) packet until instance max {b_(n),        c₁+(a_(n)−a₁)}, for n>1.

To this end, the buffer 20 needs information on W, U, b₁, c₁, b_(n) and(a_(n)−a₁). Here, W and U can be informed by the network, and b₁ andb_(n) can be easily obtained from the buffer with its own clock. Thebuffer 20 needs to keep the record of the buffer-out instance c₁. Thetime difference (a_(n)−a₁) can be calculated from the difference of thetime-stamps of the packet written at the source. As such, the buffer 20does not need to know the exact values of a_(n) or a₁, so the sourceclock does not need to be synchronized with the buffer clock.

Algorithm 1 below is an algorithm for determining a buffer holdinginterval according to an embodiment of the present disclosure.

Algorithm 1 Buffer holding interval decision 01: procedure BUFFER (m,PKT)      

 m is a preset value based on W, U, and the jitter bound                 

 W ≤ m ≤ U          

 PKT is a packet just received by the buffer 02: if first packet in theflow then 03: hold the packet with the interval (m − W)   

 The buffer-out instance of the first packet c₁ is then (b₁ + m − W) 04:TRANSMIT the packet at the decided instance 05: while there is a packetalready arrived before the first packet 06: hold the packet until theinstance max {b_(n), c₁ + (a_(n) − a₁)}             

 Let's say this is n^(th) packet, n > 1 07: TRANSMIT the packet at thedecided instance 08: end while 09: else if the first packet has alreadyarrived then 10: hold n_(th) packet until the instance max {b_(n), c₁ +(a_(n) − a₁)}             

 Let's say this is n^(th) packet, n > 1 11: TRANSMIT the packet at thedecided instance 12: else wait for another packet arrival 13: end if

The implementation of the lines 6 and 10 in Algorithm 1 is feasiblesince max {b_(n), c₁+(a_(n)−a₁)} is greater than or equal to b_(n),which is the packet departure instance of the n^(th) packet from thenetwork, by definition.

The buffer 20 has to be able to identify the first packet of a flow, inorder to identify the relative time-stamp values representing theinstance b₁ and a₁. If a flag at the header indicating that the packetis indeed the first packet is used, or if a FIFO property is guaranteedin the network, this is trivial. Alternatively, a sequence numberwritten in the packet header, such as the one in RTP, would work aswell.

If some of the earlier packets (e.g., 2^(nd) or 3^(rd) packets of aflow) arrive to the buffer sooner than the first packet, they will beheld until the first packet's buffer-out plus the additional interval,as specified in lines 5 and 6 of Algorithm 1.

FIG. 2 is a diagram showing the relationship between arrival, departureand buffer-out instances of packet of a flow under observation.

According to the present disclosure described above, it may be foundthat the following theorems hold.

Theorem 1 (Upper Bound of the End to End (E2E) Buffered Latency)

The latency from the packet arrival to the buffer-out instance(c_(n)−a_(n)) is upper bound by (m+U−W):

-   -   Proof. By definition,

$\begin{matrix}{{c_{n} - a_{n}} = {{\max\left\{ {b_{n},{c_{1} + \left( {a_{n} - a_{1}} \right)}} \right\}} - a_{n}}} \\{= {\max\left\{ {{b_{n} - a_{n}},{b_{1} + m - W - a_{1}}} \right\}}} \\{= {\max\left\{ {{b_{n} - a_{n}},{m - W + \left( {b_{1} = a_{1}} \right)}} \right\}}} \\{{{\leq {\max\left\{ {U,{m - W + U}} \right\}}} = {m - W + U}},}\end{matrix}$

since b_(n)−a_(n)≤U for any n, and m≥W.

Theorem 2 (Lower Bound of the E2E Buffered Latency)

The latency from the packet arrival to the buffer-out instance(c_(n)−a_(n)) is lower bound by m:

-   -   Proof. By definition,

$\begin{matrix}{{c_{n} - a_{n}} = {{\max\left\{ {b_{n},{c_{1} + \left( {a_{n} - a_{1}} \right)}} \right\}} - a_{n}}} \\{= {\max\left\{ {{b_{n} - a_{n}},{m - W + \left( {b_{1} - a_{1}} \right)}} \right\}}} \\{{{\geq {\max\left\{ {W,{m - W + W}} \right\}}} = m},}\end{matrix}$

since b_(n)−a_(n)≥W for any n, and m≥W.

The jitter, or the latency difference between a pair of packets, can bedefined as follows.

Definition (Jitter)

The jitter between the i^(th) packet and the j^(th) packet of a flow isdefined as follows:

r _(ij)=|(c _(i) −a _(i))−(c _(j) −a _(j))|.

Theorem 3 (Upper Bound of the Jitter)

The jitter is upper bounded by (U−m):

-   -   Proof. Define r_(n)=r_(n1)=(c_(n)−a_(n))−(c₁−a₁).    -   c₁=(b₁+m−W) and c_(n)=max {b_(n),c₁+(a_(n)−a₁)}. From the        definition,

$\begin{matrix}{r_{n} = {\left( {c_{n} - c_{1}} \right) - \left( {a_{n} - a_{1}} \right)}} \\{= {{\max\left\{ {{b_{n} - c_{1}},{a_{n} - a_{1}}} \right\}} - \left( {a_{n} - a_{1}} \right)}} \\{= {\max\left\{ {{b_{n} - c_{1} - \left( {a_{n} - a_{1}} \right)},0} \right\}}} \\{= {\max\left\{ {{b_{n} - b_{1} - m + W - \left( {a_{n} - a_{1}} \right)},0} \right\}}} \\{= {\max\left\{ {{\left( {b_{n} - a_{n}} \right) - m + W - \left( {b_{1} - a_{1}} \right)},0} \right\}}} \\{{{\leq {\max\left\{ {{U - m + W - W},0} \right\}}} = {U - m}},{{{since}\mspace{14mu} U} \geq {m.}}}\end{matrix}$

The jitter between packets i and j, r_(ij) can be rewritten such asr_(ij)=|(c_(i)−c₁)−(a_(i)−a₁)−(c_(j)−c₁)+(a_(j)−a₁)|=|r_(i)−r_(j)|.Since 0≤r_(i)≤U−m and 0≤r_(j)≤U−m, the jitter r_(ij)≤U−m, for any i,j>0.

For example, if it is assumed that there is a flow requesting theend-to-end buffered latency upper bound of 10 ms and a jitter upperbound of 1 ms, from Theorem 1, (m+U−W)=10 ms, and from Theorem 3,(U−m)=1 ms. Based on these equations, it is possible to obtain U=5.5ms+W/2 and m=4.5 ms+W/2. As such, during the call setup process, uponthe flow's requested specifications, the network and the buffer mayassign U=(5.5 ms+W/2) of the actual network latency upper bound, andm=(4.5 ms+W/2) parameter for the buffering.

As an extreme case, if it is wanted to achieve an absolutesynchronization, i.e., the inter-departure times of the output packets(c_(n)−c_(n−1)) are exactly the same as the inter-arrival times(a_(n)−a_(n−1)). Then, the jitter may be set to be equal to 0 (zero). Inthis case, the absolute synchronization may be achieved by setting m=U,the buffered latency upper bound becomes 2U−W, which is close to 2U whenW is negligible.

Hereinafter, a simple network with a small number of time-sensitiveflows will be considered to check the performance of the presentdisclosure. FIG. 3 is a diagram showing an example of a network forperformance verification of the present disclosure.

The traffic is composed of three classes with characteristics summarizedin Table 2.

TABLE 2 {Packet length, Maximum burst size, Symbol Traffic type Arrivalrate} of a flow A Audio {2 Kbit, 2 Kbit, 1.6 Mbps} V Video {12 Kbit, 360Kbit, 11 Mbps} C Command & Control {2.4 Kbit, 2.4 Kbit, 480 Kbps}

The video flow emits a larger burst compared to the other types offlows, and the number of C&C flows are more than double the number ofthe other two types of flows combined.

The flows characteristics are further simplified such that the audioflows (A flows) emit 256 byte packet every 1.25 ms, video flows (Vflows) emit 30*1500 byte bursts with 33 ms period, and C&C flows (Cflows) emit 300 byte packet every 5 ms. As shown in FIG. 3, four A flowsand one V flow share the route, so do 20 C&C flows.

Link capacities for all the links are set to be 1 Gbps, which is commonwith Gigabit Ethernet nowadays. With the notation for a flow {Packetlength, Max burst, Arrival rate}, audio flow's parameter set is {256 B,256 B, 256 B/1.25 ms} to {2 Kbit, 2 K, 1.6 Mbps}, video flows' parameterset is {1500 B, 30*1500 B, 30*1500 B/33 ms} to {12 Kbit, 360 Kbit, 11Mbps}, and C&C flows' parameter set is {300 B, 300 B, 300 B/5 ms} to{2400 bit, 2400, 480 Kbps}.

Given the topology, the input flows and their destination, three typesof solutions that guarantee latency upper bound are considered. They arethe TSN synchronous approach, the DiffServ, and the FAIR framework. Thethree types of traffic are assumed to have the same high priority.

Considering the problems to be solved by the present disclosure, theFAIR or DiffServ framework, which guarantees the latency upper bound,may be selected.

The FAIR may guarantee 0.2164 ms latency upper bound. Therefore, theparameter U equals 0.2164 ms. The sum of the packet transmission time infour nodes is 4 times of 2.4 μs, which is the latency lower bound, andthe parameter W equals to 9.6 μs. If the parameter m is set to be equalto U, 0.2164 ms, then from Theorem 1, the end-to-end buffered latencyupper bound is (m+U−W), which is equal to 0.4304 ms and the jitter upperbound is U−m. That is, a perfect synchronization can be achieved withouttime-synchronization functions specified in the TSN standard, andsimultaneously the latency upper bound requirements (10 ms in this case)can be satisfied by a wide margin, which is a much better resultcompared to TSN.

CPRI is expected to maintain the maximum end-to-end latencies less than100 to 250 μs, jitter within 65 to 100 ns, in 10 G Ethernet network. Ifscaling down to a 1 G network, it is possible to infer about 1 to 2.5 mslatency upper bound and 0.65 to 1 μs jitter upper bound.

If the jitter upper bound is set to be 1 μs, then again from Theorems 1and 3, with the FAIR framework, m=U−1 μs=0.2154 ms, m+U−W=0.4294, U−m=1μs. As such, the latency upper bound 0.4294 ms meets even the stringentCPRI requirement of 1 ms.

The DiffServ framework, or simple strict priority schedulers for highpriority traffic with preemption, can guarantee 0.8584 ms. It is alsopossible to achieve zero jitter with the DiffServ framework, with abuffer, at the cost of lengthening the end-to-end buffered latency upperbound roughly twice, to 1.7144 ms. This is the similar upper bound tothat of the TSN synchronous approach, which is 1.712 ms. With 1 μsjitter upper bound, the buffered latency upper bound is 1.7134 ms.

Hereinafter, the result of simulating the network of FIG. 3 will bedescribed. The input traffic and the routes also follow the descriptionin FIG. 3, and the flows' characteristics follow Table 2. The simulationenvironment is constructed using OMNeT++. It is a C++ based networksimulation framework, which is known to be good for constructingdiscrete event network simulation.

In the simulation, all the flows emit the maximum burst at the start ofthe simulation. It is so set in order to observe the worst case behaviorat the earlier stage of the simulation. A single run of the simulationlasts for 16.5 seconds. The C&C flows, the audio flows, and the videoflows produce 3,300, 13,200, and 15,000 packets respectively, for asingle run, and 100 runs are performed repeatedly. All C&C flows withthe same input port and output port were observed, and the result wasobtained using the latency observed for 6,600,000 packets.

FIG. 4 is a diagram showing a latency distribution obtained from flowsimulation of the C&C using end-to-end latency upper bound and threedifferent frameworks.

Referring to FIG. 4, the theoretical latency upper bound in allframeworks matches the maximum observed latencies. In the TAS framework,the worst latency is observed when the slots of the three types of flowoverlap. In addition, the middle dot that looks like a short line in theTAS indicates the packet of the slot that partially overlaps the slot ofthe video flow.

FIG. 5 is a diagram showing end-to-end buffering latency of all packetsin two arbitrary sample C&C flows observed during the simulation.

Referring to FIG. 5, the latency of a subsequent packet is determinedaccording to the actual latency experienced by the first packet of theflow. The fixed latency of the sample flow is located near theend-to-end latency upper bound of 0.2164 ms. It was confirmed that theflow achieved zero latency.

While the present disclosure has been described with reference to theembodiments illustrated in the figures, the embodiments are merelyexamples, and it will be understood by those skilled in the art thatvarious changes in form and other embodiments equivalent thereto can beperformed. Therefore, the technical scope of the disclosure is definedby the technical idea of the appended claims The drawings and theforgoing description gave examples of the present invention. The scopeof the present invention, however, is by no means limited by thesespecific examples. Numerous variations, whether explicitly given in thespecification or not, such as differences in structure, dimension, anduse of material, are possible. The scope of the invention is at least asbroad as given by the following claims.

What is claimed is:
 1. A method of guaranteeing a jitter upper bound fora network without time-synchronization, which guarantees a jitter upperbound for a flow that is transmitted from a source to a destinationthrough a network, setting a jitter upper bound to an arbitrary value,determining a predetermined buffer holding interval, holding, at abuffer located between the network and the destination, a packet of theflow for the predetermined holding interval, the buffer outputting thepacket of the flow, and wherein the network guarantees a latency upperbound of the flow.
 2. The method of guaranteeing a jitter upper boundfor a network without time-synchronization of claim 1, wherein thearbitrary value can be as low as 0 (zero).
 3. The method of guaranteeinga jitter upper bound for a network without time-synchronization of claim2, wherein the packet includes a relative time-stamp in which adeparture instance from the source and an arrival instance to thenetwork are recorded, and wherein the buffer holding interval isdetermined using the relative time-stamp.
 4. The method of guaranteeinga jitter upper bound for a network without time-synchronization of claim3, wherein the buffer is configured to: hold a first packet of the flowfor a buffer holding interval (m−W); and hold a n^(th) packet of theflow until an instance max {b_(n), c₁+(a_(n)−a₁)}, where a_(n) is anarrival instance when the n^(th) packet of the flow arrives at thenetwork, b_(n) is a departure instance when the n^(th) packet departsfrom the network, c_(n) is a buffer-out instance when the n^(th) packetdeparts from the buffer, U is the latency upper bound, W is a latencylower bound provided by the network, m is a buffering parameter, W≤m≤U,and n>1.
 5. The method of guaranteeing a jitter upper bound for anetwork without time-synchronization of claim 4, wherein the jitterupper bound is U−m, and zero jitter is implemented by setting m=U. 6.The method of guaranteeing a jitter upper bound for a network withouttime-synchronization of claim 5, wherein the arbitrary value is set to 0(zero).
 7. A method of guaranteeing a jitter upper bound for a networkwithout time-synchronization, which guarantees a jitter upper bound fora flow that is transmitted from a source to a destination through anetwork, setting a jitter upper bound to an arbitrary value, determininga predetermined buffer holding interval, holding, at a buffer located ata boundary of the network, a packet of the flow for the predeterminedholding interval, the buffer outputting the packet of the flow, andwherein the network guarantees a latency upper bound of the flow.
 8. Themethod of guaranteeing a jitter upper bound for a network withouttime-synchronization of claim 7, wherein the arbitrary value can be aslow as 0 (zero).
 9. The method of guaranteeing a jitter upper bound fora network without time-synchronization of claim 8, wherein the packetincludes a relative time-stamp in which a departure instance from thesource and an arrival instance to the network are recorded, and whereinthe buffer holding interval is determined using the relative time-stamp.10. The method of guaranteeing a jitter upper bound for a networkwithout time-synchronization of claim 9, wherein the buffer isconfigured to: hold a first packet of the flow for a buffer holdinginterval (m−W); and hold a n^(th) packet of the flow until an instancemax {b_(n), c₁+(a_(n)−a₁)}, where a_(n) is an arrival instance when then^(th) packet of the flow arrives at the network, b_(n) is a departureinstance when the n^(th) packet departs from the network, c_(n) is abuffer-out instance when the n^(th) packet departs from the buffer, U isthe latency upper bound, W is a latency lower bound provided by thenetwork, m is a buffering parameter, W≤m≤U, and n>1.
 11. The method ofguaranteeing a jitter upper bound for a network withouttime-synchronization of claim 10, wherein the jitter upper bound is U−m,and zero jitter is implemented by setting m=U.
 12. The method ofguaranteeing a jitter upper bound for a network withouttime-synchronization of claim 11, wherein the arbitrary value is set to0 (zero).